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mod_mariadb install problem
by Mehdi Shirazi 30 Jun '20

30 Jun '20

Hi I tried to install mod_mariadb in debian 9 Freeswitch 1.10.3 and mariadb 10.4 via package. When I want to load start module I see this error: 2020-06-20 12:02:15.274711 [CRIT] switch_loadable_module.c:1785 Error Loading module /usr/lib/freeswitch/mod/mod_mariadb.so **/usr/lib/freeswitch/mod/mod_mariadb.so: symbol mysql_get_timeout_value, version libmysqlclient_18 not defined in file libmariadb.so.3 with link time reference** please help me solve the problem. Regards M.Shirazi

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Incoming call audio issue
by Roberto Scarpone 30 Jun '20

30 Jun '20

Hi, I have a problem when phones receive a call, for a time varying between 1-3 seconds, they hear nothing (at the beginning of the call). The problem does not always occur, but with a frequency of 2/3 time every 10 calls. After a few seconds the conversation presents no further problems. Listening to the captured audio the call doesn't seems to have any problem, the audio is complete, but when they answer the call the problem is there and they can't hear the first 1-3 seconds of the conversation. I have a FreeSWITCH 1.6.20 on Debian 8, installed from repo. Has anyone already encountered this problem or would you know how to help me solve it? Best regards, Roberto

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UniMRCP Setting variables
by David Villasmil 29 Jun '20

29 Jun '20

Hello, I'm testing unimrcp server with freeswitch using mod_unimrcp. Is there any way to set channel variables on the unimrcp channel? i.e.: I need to set some variables on the recognition channel, like all of the incoming channel variables. Thanks! David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337

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PBX Call-Transfer Issue - timing of RTP?
by Andy Newlands 29 Jun '20

29 Jun '20

We have Freeswitch acting as a SIP platform, with internects with various UK carriers (including BT) on one side, with customer PBX connected from the other side. We have an intermittent problem where a call comes into FS from the PSTN and is routed to the customer PBX (and Avaya), with the call then transferred back, via Freeswitch, then the PSTN to another phone number. Sometimes the original caller continues to hear ringtone after the transfer-destination has answered (and no conversation is possible). I have attached Wireshark SIP flows for the scenario where this works and the one where transfer does not complete properly (labelled in red, above each ladder diagram, "BROKEN", on the left and "WORKING" on the right). In the working example, we see RTP BEFORE the REFER. In the non-working example, RTP is sent AFTER the BYE. 1. We don't understand why the Avaya sends messages in a different order, on different occasions, for the exact same operation? 2. Should Freeswitch be able to handle both the scenarios shown in the attached image? 3. Are there any config or dialplan settings which can be used to help fix this? NOTE: We also have other issues with call transfers in general, whereby the transfer succeeds, but there is no audio heard by the transferee. Thanks

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Crypto negotiated, but FS thinks otherwise and drops outbound calls.
by Keith Laaks 29 Jun '20

29 Jun '20

> Hi All, > > I am running: FreeSWITCH Version 1.10.3-release-15-129de34d84~64bit (-release-15-129de34d84 64bit) on Debian GNU/Linux 10 (buster) > > I am bridging an incoming call to a destination over SIP TLS, using SRTP and ICE-lite. rtcp_mux is disabled. > > I have to use ‘mandatory’ in the export below (before the bridge), as this specific destination fails the call if the SDP includes the unencrypted optioN (i.e. m=audio 2354 RTP/AVP 18 8 98) > <action application="export" data=“rtp_secure_media=mandatory:AES_CM_128_HMAC_SHA1_32"/> > > I am having a problem in that FS drops the call as soon as the destination answers. > > In the debug traces I can see that, although Secure RTP is indeed being negotiated (Activating audio Secure RTP RECV) FS somehow thinks otherwise (switch_core_media.c:5665 Crypto not negotiated but required.) > > 2020-06-26 18:20:40.208635 [INFO] switch_core_media.c:8843 Activating Audio ICE > 2020-06-26 18:20:40.208635 [NOTICE] switch_rtp.c:4915 Activating RTP audio ICE: ZyVg:nzPxIzWUEUYj6GjG 11.22.33.44:50004 > 2020-06-26 18:20:40.208635 [DEBUG] switch_core_media.c:8975 sofia/teams/27101234567 Set 2833 dtmf send payload to 101 > 2020-06-26 18:20:40.208635 [DEBUG] switch_core_media.c:8982 sofia/teams/27101234567 Set 2833 dtmf receive payload to 101 > 2020-06-26 18:20:40.208635 [DEBUG] switch_core_media.c:9005 sofia/teams/27101234567 Set rtp dtmf delay to 40 > 2020-06-26 18:20:40.208635 [DEBUG] switch_core_media.c:1501 LIFETIME found in |2^31, base 2 exp 31 > 2020-06-26 18:20:40.208635 [NOTICE] switch_core_media.c:1524 Skipping MKI due to empty index > 2020-06-26 18:20:40.208635 [INFO] switch_rtp.c:4217 Activating audio Secure RTP SEND > 2020-06-26 18:20:40.208635 [INFO] switch_rtp.c:4195 Activating audio Secure RTP RECV > 2020-06-26 18:20:40.208635 [DEBUG] switch_core_sqldb.c:2836 Secure Type: srtp:sdes:AES_CM_128_HMAC_SHA1_32 > 2020-06-26 18:20:40.208635 [DEBUG] switch_core_sqldb.c:2836 Secure Type: srtp:sdes:AES_CM_128_HMAC_SHA1_32 > 2020-06-26 18:20:40.208635 [NOTICE] sofia.c:8503 Channel [sofia/teams/27101234567] has been answered > 2020-06-26 18:20:40.208635 [DEBUG] switch_channel.c:3865 (sofia/teams/27101234567) Callstate Change RINGING -> ACTIVE > 2020-06-26 18:20:40.228625 [DEBUG] switch_core_media.c:5592 Audio Codec Compare [G729:18:8000:20:8000:1]/[opus:116:48000:20:0:1] > 2020-06-26 18:20:40.228625 [DEBUG] switch_core_media.c:5592 Audio Codec Compare [G729:18:8000:20:8000:1]/[PCMA:8:8000:20:64000:1] > 2020-06-26 18:20:40.228625 [DEBUG] switch_core_media.c:5592 Audio Codec Compare [G729:18:8000:20:8000:1]/[PCMU:0:8000:20:64000:1] > 2020-06-26 18:20:40.228625 [DEBUG] switch_core_media.c:5592 Audio Codec Compare [PCMA:8:8000:20:64000:1]/[opus:116:48000:20:0:1] > 2020-06-26 18:20:40.228625 [DEBUG] switch_core_media.c:5592 Audio Codec Compare [PCMA:8:8000:20:64000:1]/[PCMA:8:8000:20:64000:1] > 2020-06-26 18:20:40.228625 [DEBUG] switch_core_media.c:5647 Audio Codec Compare [PCMA:8:8000:20:64000:1] ++++ is saved as a match > 2020-06-26 18:20:40.228625 [DEBUG] switch_core_media.c:5592 Audio Codec Compare [PCMA:8:8000:20:64000:1]/[PCMU:0:8000:20:64000:1] > 2020-06-26 18:20:40.228625 [DEBUG] switch_core_media.c:5508 Set telephone-event payload to 98@8000 > 2020-06-26 18:20:40.228625 [WARNING] switch_core_media.c:5665 Crypto not negotiated but required. > 2020-06-26 18:20:40.228625 [DEBUG] switch_core_media.c:5909 sofia/vox/0831233210(a)10.10.1.1 <mailto:sofia/vox/0831233210@10.10.1.1> Set 2833 dtmf send payload to 98 recv payload to 98 > 2020-06-26 18:20:40.228625 [NOTICE] switch_channel.c:3908 Hangup sofia/vox/0831233210(a)10.10.1.1 <mailto:sofia/vox/0831233210@10.10.1.1> [CS_EXECUTE] [INCOMPATIBLE_DESTINATION] > > Am I missing something here? > > Thanks > > Keith >

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Compilation of Vanilla FreeSWITCH on Centos 7
by Kumar, Puneet (Puneet) 29 Jun '20

29 Jun '20

Hello, I followed steps given at link https://freeswitch.org/confluence/display/FREESWITCH/CentOS+7+and+RHEL+7 to compile FreeSWITCH v1.10.3 code. But getting following errors on make : CC libavmod_la-mod_av.lo CC libavmod_la-avformat.lo In file included from /usr/include/ffmpeg/libavutil/imgutils.h:31:0, from avformat.c:39: /usr/include/ffmpeg/libavutil/pixdesc.h:34:5: error: type of bit-field 'plane' is a GCC extension [-Werror=pedantic] uint16_t plane : 2; ^ /usr/include/ffmpeg/libavutil/pixdesc.h:40:5: error: type of bit-field 'step_minus1' is a GCC extension [-Werror=pedantic] uint16_t step_minus1 : 3; ^ /usr/include/ffmpeg/libavutil/pixdesc.h:46:5: error: type of bit-field 'offset_plus1' is a GCC extension [-Werror=pedantic] uint16_t offset_plus1 : 3; ^ /usr/include/ffmpeg/libavutil/pixdesc.h:52:5: error: type of bit-field 'shift' is a GCC extension [-Werror=pedantic] uint16_t shift : 3; ^ /usr/include/ffmpeg/libavutil/pixdesc.h:57:5: error: type of bit-field 'depth_minus1' is a GCC extension [-Werror=pedantic] uint16_t depth_minus1 : 4; ^ avformat.c:44:38: fatal error: libavresample/avresample.h: No such file or directory #include <libavresample/avresample.h> ^ cc1: all warnings being treated as errors compilation terminated. Am I missing something? Thanks, Puneet

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Drop A leg and originate a B-leg
by David Villasmil 28 Jun '20

28 Jun '20

Hello all, So right now I'm generating A simple INVITE, let's say from FS-G. On receiving the INVITE, FS launches a python script which should reject the incoming INVITE from G and start a new call B-leg. I already do this, but the FS-G leg keeps going and I would like to actually reject and hangup the FS-G leg. is this possible? Basically hangup the incoming call and keep the script running to create a new leg. Regards, David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337

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OSError when using Python ESL + socketserver
by René Weiss 26 Jun '20

26 Jun '20

Hi, When I'm using the ESL library (python-ESL==1.4.18) for Python (3.7.7) together with socketserver I always get an OSError when the connection is closing. Here some sample code (mostly the example from the wiki ported to Python 3): ----------- import socketserver import ESL class ESLReqHandler(socketserver.BaseRequestHandler): def setup(self): print(self.client_address, 'connected!') fd = self.request.fileno() print(fd) con = ESL.ESLconnection(fd) print('Connected: ', con.connected()) if con.connected(): info = con.getInfo() uuid = info.getHeader("unique-id") print(uuid) con.execute("answer", "", uuid) con.execute("sleep", "2000", uuid) server = socketserver.ThreadingTCPServer(('', 8040), ESLReqHandler) server.serve_forever() ----------- And this is the error message I get: ----------- Exception in thread Thread-1: Traceback (most recent call last): File "/usr/local/Cellar/python/3.7.7/Frameworks/Python.framework/Versions/3.7/lib/python3.7/threading.py", line 926, in _bootstrap_inner self.run() File "/usr/local/Cellar/python/3.7.7/Frameworks/Python.framework/Versions/3.7/lib/python3.7/threading.py", line 870, in run self._target(*self._args, **self._kwargs) File "/usr/local/Cellar/python/3.7.7/Frameworks/Python.framework/Versions/3.7/lib/python3.7/socketserver.py", line 654, in process_request_thread self.shutdown_request(request) File "/usr/local/Cellar/python/3.7.7/Frameworks/Python.framework/Versions/3.7/lib/python3.7/socketserver.py", line 509, in shutdown_request self.close_request(request) File "/usr/local/Cellar/python/3.7.7/Frameworks/Python.framework/Versions/3.7/lib/python3.7/socketserver.py", line 513, in close_request request.close() File "/usr/local/Cellar/python/3.7.7/Frameworks/Python.framework/Versions/3.7/lib/python3.7/socket.py", line 420, in close self._real_close() File "/usr/local/Cellar/python/3.7.7/Frameworks/Python.framework/Versions/3.7/lib/python3.7/socket.py", line 414, in _real_close _ss.close(self) OSError: [Errno 9] Bad file descriptor ----------- The problem starts to appear as soon as the command "ESL.ESLconnection(fd)" is included in the setup function. Any ideas? Thanks René Weiss

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Noise on line, does not show up in recordings
by Sam van Herwaarden 26 Jun '20

26 Jun '20

Hi all, I frequently get noise during calls with my FreeSWITCH deployment. It sounds like a brief burst of distortion - typically less than a second. I've had it happen both when using playback with WAV (different formats - I found recommendations suggesting 8000Hz, 128 kb/s mono and it shows up with that format as well) and text-to-speech with flite. It happens at unpredictable moments - it's not the same from one call to the next. I've added a record_session statement to investigate in more detail, but the noise does not show up in the recording. Does anyone have advice on what might be causing this or how I could further debug this? Kind regards, Sam

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http_cache upload failure
by David Villasmil 25 Jun '20

25 Jun '20

Hello all, I'm doing: <action application="record" data="http_cache:// https://mybucket.s3.amazonaws.com/file.mp3 30 3 2"/> and i'm getting 2020-06-25 02:14:18.251194 [NOTICE] sofia.c:1079 Hangup sofia/external/ vozelia-prod-1@37.139.121.39:5080 [CS_EXECUTE] [NORMAL_CLEARING] 2020-06-25 02:14:18.251194 [DEBUG] mod_http_cache.c:342 opening /var/cache/freeswitch/b6/349a08-37ac-409d-b607-0c0b72369d02.mp3 for upload to https://mybucket.s3.amazonaws.com/file.mp3 2020-06-25 02:14:18.291196 [ERR] mod_http_cache.c:431 Received HTTP error 0 trying to save /var/cache/freeswitch/b6/349a08-37ac-409d-b607-0c0b72369d02.mp3 to https://mybucket.s3.amazonaws.com/file.mp3 but i can upload from the CLI curl -X PUT -T "ef8d02-5738-4c9b-886f-934e8a5fc051.mp3" https://mybucket.s3.amazonaws.com/file.mp3 Works perfectly... Any ideas? Many thanks! Regards, David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337

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